michael@0: /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
michael@0: /* vim:set ts=2 sw=2 sts=2 et cindent: */
michael@0: /* This Source Code Form is subject to the terms of the Mozilla Public
michael@0:  * License, v. 2.0. If a copy of the MPL was not distributed with this
michael@0:  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
michael@0: #ifndef MOZILLA_AUDIOSAMPLEFORMAT_H_
michael@0: #define MOZILLA_AUDIOSAMPLEFORMAT_H_
michael@0: 
michael@0: #include "nsAlgorithm.h"
michael@0: #include <algorithm>
michael@0: 
michael@0: namespace mozilla {
michael@0: 
michael@0: /**
michael@0:  * Audio formats supported in MediaStreams and media elements.
michael@0:  *
michael@0:  * Only one of these is supported by AudioStream, and that is determined
michael@0:  * at compile time (roughly, FLOAT32 on desktops, S16 on mobile). Media decoders
michael@0:  * produce that format only; queued AudioData always uses that format.
michael@0:  */
michael@0: enum AudioSampleFormat
michael@0: {
michael@0:   // Native-endian signed 16-bit audio samples
michael@0:   AUDIO_FORMAT_S16,
michael@0:   // Signed 32-bit float samples
michael@0:   AUDIO_FORMAT_FLOAT32,
michael@0:   // Silence: format will be chosen later
michael@0:   AUDIO_FORMAT_SILENCE,
michael@0:   // The format used for output by AudioStream.
michael@0: #ifdef MOZ_SAMPLE_TYPE_S16
michael@0:   AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_S16
michael@0: #else
michael@0:   AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_FLOAT32
michael@0: #endif
michael@0: };
michael@0: 
michael@0: enum {
michael@0:   MAX_AUDIO_SAMPLE_SIZE = sizeof(float)
michael@0: };
michael@0: 
michael@0: template <AudioSampleFormat Format> class AudioSampleTraits;
michael@0: 
michael@0: template <> class AudioSampleTraits<AUDIO_FORMAT_FLOAT32> {
michael@0: public:
michael@0:   typedef float Type;
michael@0: };
michael@0: template <> class AudioSampleTraits<AUDIO_FORMAT_S16> {
michael@0: public:
michael@0:   typedef int16_t Type;
michael@0: };
michael@0: 
michael@0: typedef AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type AudioDataValue;
michael@0: 
michael@0: template<typename T> class AudioSampleTypeToFormat;
michael@0: 
michael@0: template <> class AudioSampleTypeToFormat<float> {
michael@0: public:
michael@0:   static const AudioSampleFormat Format = AUDIO_FORMAT_FLOAT32;
michael@0: };
michael@0: 
michael@0: template <> class AudioSampleTypeToFormat<short> {
michael@0: public:
michael@0:   static const AudioSampleFormat Format = AUDIO_FORMAT_S16;
michael@0: };
michael@0: 
michael@0: // Single-sample conversion
michael@0: /*
michael@0:  * Use "2^N" conversion since it's simple, fast, "bit transparent", used by
michael@0:  * many other libraries and apparently behaves reasonably.
michael@0:  * http://blog.bjornroche.com/2009/12/int-float-int-its-jungle-out-there.html
michael@0:  * http://blog.bjornroche.com/2009/12/linearity-and-dynamic-range-in-int.html
michael@0:  */
michael@0: inline float
michael@0: AudioSampleToFloat(float aValue)
michael@0: {
michael@0:   return aValue;
michael@0: }
michael@0: inline float
michael@0: AudioSampleToFloat(int16_t aValue)
michael@0: {
michael@0:   return aValue/32768.0f;
michael@0: }
michael@0: 
michael@0: template <typename T> T FloatToAudioSample(float aValue);
michael@0: 
michael@0: template <> inline float
michael@0: FloatToAudioSample<float>(float aValue)
michael@0: {
michael@0:   return aValue;
michael@0: }
michael@0: template <> inline int16_t
michael@0: FloatToAudioSample<int16_t>(float aValue)
michael@0: {
michael@0:   float v = aValue*32768.0f;
michael@0:   float clamped = std::max(-32768.0f, std::min(32767.0f, v));
michael@0:   return int16_t(clamped);
michael@0: }
michael@0: 
michael@0: // Sample buffer conversion
michael@0: 
michael@0: template <typename From, typename To> inline void
michael@0: ConvertAudioSamples(const From* aFrom, To* aTo, int aCount)
michael@0: {
michael@0:   for (int i = 0; i < aCount; ++i) {
michael@0:     aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i]));
michael@0:   }
michael@0: }
michael@0: inline void
michael@0: ConvertAudioSamples(const int16_t* aFrom, int16_t* aTo, int aCount)
michael@0: {
michael@0:   memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
michael@0: }
michael@0: inline void
michael@0: ConvertAudioSamples(const float* aFrom, float* aTo, int aCount)
michael@0: {
michael@0:   memcpy(aTo, aFrom, sizeof(*aTo)*aCount);
michael@0: }
michael@0: 
michael@0: // Sample buffer conversion with scale
michael@0: 
michael@0: template <typename From, typename To> inline void
michael@0: ConvertAudioSamplesWithScale(const From* aFrom, To* aTo, int aCount, float aScale)
michael@0: {
michael@0:   if (aScale == 1.0f) {
michael@0:     ConvertAudioSamples(aFrom, aTo, aCount);
michael@0:     return;
michael@0:   }
michael@0:   for (int i = 0; i < aCount; ++i) {
michael@0:     aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i])*aScale);
michael@0:   }
michael@0: }
michael@0: inline void
michael@0: ConvertAudioSamplesWithScale(const int16_t* aFrom, int16_t* aTo, int aCount, float aScale)
michael@0: {
michael@0:   if (aScale == 1.0f) {
michael@0:     ConvertAudioSamples(aFrom, aTo, aCount);
michael@0:     return;
michael@0:   }
michael@0:   if (0.0f <= aScale && aScale < 1.0f) {
michael@0:     int32_t scale = int32_t((1 << 16) * aScale);
michael@0:     for (int i = 0; i < aCount; ++i) {
michael@0:       aTo[i] = int16_t((int32_t(aFrom[i]) * scale) >> 16);
michael@0:     }
michael@0:     return;
michael@0:   }
michael@0:   for (int i = 0; i < aCount; ++i) {
michael@0:     aTo[i] = FloatToAudioSample<int16_t>(AudioSampleToFloat(aFrom[i])*aScale);
michael@0:   }
michael@0: }
michael@0: 
michael@0: // In place audio sample scaling.
michael@0: inline void
michael@0: ScaleAudioSamples(float* aBuffer, int aCount, float aScale)
michael@0: {
michael@0:   for (int32_t i = 0; i < aCount; ++i) {
michael@0:     aBuffer[i] *= aScale;
michael@0:   }
michael@0: }
michael@0: 
michael@0: inline void
michael@0: ScaleAudioSamples(short* aBuffer, int aCount, float aScale)
michael@0: {
michael@0:   int32_t volume = int32_t((1 << 16) * aScale);
michael@0:   for (int32_t i = 0; i < aCount; ++i) {
michael@0:     aBuffer[i] = short((int32_t(aBuffer[i]) * volume) >> 16);
michael@0:   }
michael@0: }
michael@0: 
michael@0: inline const void*
michael@0: AddAudioSampleOffset(const void* aBase, AudioSampleFormat aFormat,
michael@0:                      int32_t aOffset)
michael@0: {
michael@0:   static_assert(AUDIO_FORMAT_S16 == 0, "Bad constant");
michael@0:   static_assert(AUDIO_FORMAT_FLOAT32 == 1, "Bad constant");
michael@0:   NS_ASSERTION(aFormat == AUDIO_FORMAT_S16 || aFormat == AUDIO_FORMAT_FLOAT32,
michael@0:                "Unknown format");
michael@0: 
michael@0:   return static_cast<const uint8_t*>(aBase) + (aFormat + 1)*2*aOffset;
michael@0: }
michael@0: 
michael@0: } // namespace mozilla
michael@0: 
michael@0: #endif /* MOZILLA_AUDIOSAMPLEFORMAT_H_ */